Regardless of public line or intracompany communications, it is a problem of communications to reduce communication cost and increasing availability of the line. To achieve high efficiency transmission of speech signals forming a great majority of communication traffic, a variety of speech encoding schemes are implemented. Recently, digital line transmission units have been implemented which utilize high efficiency speech encoding units based on low bit rate speech encoding/decoding schemes typified by an 8 kbit/s CS-ACELP (Conjugate-Structure Algebraic-Code-Excited Linear Prediction) speech encoding scheme.
To make the best possible use of a line, a more efficient speech encoding unit (called “speech codec” from now on) must be employed. In other words, a lower rate speech encoding unit must be applied. Consider a case where a low-bit-rate speech codec is applied. For example, applying an 8-kbit/s speech codec to a 64-kbit/s speech PCM signal can increase the compression efficiency by a factor of eight. In contrast, applying 32-kbit/s speech codec can achieve compression efficiency by a factor of two.
Although some low-bit-rate encoding/decoding schemes of the speech codecs achieve high sound quality, it is obvious that high-bit-rate codecs generally bring about higher sound quality than low-bit-rate codecs.
Accordingly, the concept has been developed that when a communication line allows a margin for information transmission, a highest-possible-bit-rate codec is used to secure sound quality, whereas when it leaves little margin for information transmission, a low-bit-rate codec is used to make the best possible use of the communication line. This type includes a unit that varies the bit rate of the speech codec used for the speech encoding/decoding depending on the line condition by applying a variable rate codec, and a unit that switches the codecs in accordance with the traffic volume on the communication line.
FIG. 1 is a block diagram showing a configuration of a conventional digital line transmission unit that switches the codecs, which is disclosed in Japanese patent application laid-open No. 63-226138/1988, for example. In FIG. 1, the reference numeral 40 designates a transmitting side transmission unit, and 45 designates an input line to which a speech signal is input. The reference numeral 46 designates a 64-kbit/s codec for receiving the speech signal and outputting an encoded speech signal; 47 designates a 32-kbit/s codec for receiving the speech signal and outputting an encoded speech signal; and 48 designates a 16-kbit/s codec for receiving the speech signal and outputting a speech compression signal. The reference numeral 49 designates a selector for receiving the speech compression signals from these codecs and outputting a selected speech compression signal. The reference numeral 50 designates a multiplexing/demultiplexing circuit for receiving the selected speech compression signal and a control signal and outputting a multiplexed signal. The reference numeral 51 designates a control circuit for outputting a control signal. The reference numeral 60 designates a receiving side transmission unit, in which the reference numeral 66 designates a 64-kbit/s codec for receiving a speech compression signal and outputting a speech signal, 67 designates a 32-kbit/s codec for receiving the speech compression signal and outputting the speech signal, and 68 designates a 16-kbit/s codec for receiving the speech compression signal and outputting the speech signal. The reference numeral 69 designates a selector for receiving the speech compression signal and supplying it to the codecs. The reference numeral 70 designates a multiplexing/demultiplexing circuit for receiving the multiplexed signal and outputting the speech compression signal and control information signal. The reference numeral 71 designates a control circuit for receiving the control information signal and outputting the control signal. The reference numeral 75 designates an output line for outputting the speech signal. The reference numeral 52 designates a digital line for interconnecting the transmitting side transmission unit 40 and the receiving side transmission unit 60.
Next, the operation will be described.
The speech signal input to the input line 45 is supplied to the 64-kbit/s codec 46, 32-kbit/s codec 47 and 16-kbit/s codec 48, which encode the speech signal and produce the speech compression signals. The speech compression signals the codecs output are supplied to the selector 49. The control circuit 51 detects the traffic on the digital line 52 before the speech signal is input to the individual codecs. To select the optimum speech compression signal for the current traffic volume, the control circuit 51 selects the codec to be used for the communication between the transmitting and receiving digital line transmission units from among the codecs 46, 47 and 48 in advance.
As for the transmission of the control information such as on selection and decision of the codec between the transmitting side transmission unit 40 and receiving side transmission unit 60, it is carried out between the control circuit 51 of the transmitting side transmission unit 40 and the control circuit 71 of the receiving side transmission unit 60 through a control information signal channel (shown in FIG. 2). FIG. 2 shows an example of a communication format. In this example, a control information signal channel S provided at the edge of information channels CH1, CH2, . . . , and CHn is used.
The selector 49 selects the codec with the optimum compression ratio for the telephone conversation in response to the control signal from the control circuit 51, and supplies the speech compression signal from the selected codec to the multiplexing/demultiplexing circuit 50. The multiplexing/demultiplexing circuit 50 multiplexes the digital signal corresponding to the speech compression which is selected and decided in response to the control signal from the control circuit 51, and transmits it through the digital line 52. The multiplexing/demultiplexing circuit 50 carries out the multiplexing and demultiplexing not only of the speech compression signal, but also of the control information signal between the control circuits 51 and 71.
In the receiving side transmission unit 60, the multiplexing/demultiplexing circuit 70 demultiplexes the speech compression signal (encoded speech signal), which is supplied to the selector 69. The encoded speech signal is supplied to the codec which is selected and decided in response to the control signal fed from the control circuit 71. The 64-kbit/s codec 66, 32-kbit/s codec 67 and 16-kbit/s codec 68 decode the encoded speech signal fed from the selector 69 to generate the speech signal. The speech signal is output in accordance with the designation of the control signal about the codec output by the control circuit 71.
When the codec is switched in the same call, the conventional digital line transmission unit involves a processing delay difference in the codecs based on different encoding/decoding schemes. Thus, the switching will bring about omission or overlap of the speech signal, giving an unnatural feeling to the auditory perception. To avoid this, the switching of the codec to be used is made on a call by call basis. More specifically, it checks inherent information on the control between adjacent multiplexing transmission units, which is transmitted through the control information signal channel as shown in FIG. 2, decides the codec to be used in accordance with the line condition in advance by the time an outgoing call processing or an incoming call processing is completed, followed by the transmission or reception (that is, by the time the call is established), and uses the same codec until the call has been completed.
With the foregoing configuration, the conventional digital line transmission unit has a problem in that since it cannot switch the codec while a call holds, it cannot increase the efficiency in terms of making effective use of the line.
The present invention is implemented to solve the foregoing problems. Therefore an object of the present invention is to provide a digital line transmission unit capable of giving a natural feeling to the auditory perception in spite of the switching between the speech codecs in accordance with the condition of the communication line, and capable of enabling effective use of the communication line without fail.